GB_Record_Node * GB_Find_Record_Node_by_Call_ID(GB_CONNECT_STATE *gb_cons, osip_call_id_t *call_id, int *index)
{
GB_Record_Node *record = NULL;
int pos;
int list_size = 0;
if(gb_cons == NULL || call_id == NULL)
{
TRACE(SCI_TRACE_NORMAL,MOD_GB,"%s line=%d Err\n",__FUNCTION__,__LINE__);
return NULL;
}
list_size = osip_list_size(&(gb_cons->record_node_list));
if(list_size <= 0)
{
return NULL;
}
for(pos=0; pos<list_size; pos++)
{
record = osip_list_get(&(gb_cons->record_node_list), pos);
if(record)
{
if(osip_call_id_match(record->call_id, call_id) == 0)
{
*index = pos;
return record;
}
record = NULL;
}
}
return NULL;
}
static SalOp * sal_find_in_subscribe_by_call_id(Sal *sal, osip_call_id_t *call_id){
const MSList *elem;
SalOp *op;
for(elem=sal->in_subscribes;elem!=NULL;elem=elem->next){
op=(SalOp*)elem->data;
if (op->call_id && osip_call_id_match(op->call_id,call_id)==0)
return op;
}
return NULL;
}
static struct session *sip_callback(struct session *sip, u_char *data, uint32_t len)
{
osip_message_t *msg;
struct tuple4 addr;
struct session *start;
struct session *rtp;
osip_call_id_t *call_id;
osip_message_init(&msg);
if (!osip_message_parse(msg, (char *)data, len)) {
if (NULL == sip->u.sip_params.call_id) {
/*
* If the session object was created by udp_callback
* we need to fill in the call_id field here because
* udp_callback doesn't know anything about SIP
*/
if (NULL != (call_id = osip_message_get_call_id(msg)))
osip_call_id_clone(call_id, &sip->u.sip_params.call_id);
} else {
/*
* Otherwise check if the session object passed to this
* function call was really the one corresponding to the
* call ID in the SIP packet, in case several SIP calls
* are passed upon the same transport layer protocol,
* source and destination IPv4 address & port combination.
* udp_callback has no way of knowing how to distinguish
* SIP session objects based on call ID, so we have to do
* it here. We just continue searching in the list of all
* tracked sessions for similar SIP objects until we find
* one that has the same call ID, or else we create a new
* SIP session object that corresponds to the new call.
*/
start = sip;
do {
if (NULL == (call_id = osip_message_get_call_id(msg)))
continue;
if (!osip_call_id_match(sip->u.sip_params.call_id, call_id))
break;
} while ((sip = sessions_find(sip->next, TYPE_SIP, 0, &sip->addr)));
if (NULL == sip) {
if (bonus_time) {
osip_message_free(msg);
return start;
}
sip = sessions_add(start->type, &start->addr, NULL);
if (NULL != (call_id = osip_message_get_call_id(msg)))
osip_call_id_clone(call_id, &sip->u.sip_params.call_id);
}
}
/*
* If the current SIP packet is an INVITE message, store the
* advertised source port and IPv4 address. It is not very
* important, since we can do only with the destination part
* (useful in case the capture missed the INVITE packet), but
* it helps discriminating from unrelated packets.
*
* Unfortunately, some SIP implementations such as the one in
* Audiocodes Mediant 1000 SIP gateways actually use a source
* port different from the one they advertised in the INVITE
* message parameters - how outrageous! - so we have to make
* our sessions search engine ignore the source port part by
* zeroing it :-/
*/
if (MSG_IS_INVITE(msg)) {
if (!bonus_time) {
sip_get_address(msg, &sip->u.sip_params.rtp_addr.saddr, &sip->u.sip_params.rtp_addr.source);
#ifndef USING_NON_STUPID_SIP_IMPLEMENTATIONS
sip->u.sip_params.rtp_addr.source = 0;
#endif
}
} else
if (MSG_TEST_CODE(msg, 200)) {
if (MSG_IS_RESPONSE_FOR(msg, "INVITE")) {
if (!bonus_time && sip_get_address(msg, &sip->u.sip_params.rtp_addr.daddr, &sip->u.sip_params.rtp_addr.dest)) {
sessions_add(TYPE_UDP | TYPE_RTP, &sip->u.sip_params.rtp_addr, sip);
sip->u.sip_params.picked_up = 1;
}
} else
if (MSG_IS_RESPONSE_FOR(msg, "BYE") ||
MSG_IS_RESPONSE_FOR(msg, "CANCEL")) {
start = first_session;
while (NULL != (rtp = sessions_find(start, TYPE_RTP, sip->id, NULL))) {
sessions_del(rtp);
start = rtp->next;
}
/*
* Mark for deletion in 2 seconds, in order to give some
* time to the extra ACK packets that might be exchanged
*/
if (sip->type & TYPE_UDP)
sip->timeout = nids_last_pcap_header->ts.tv_sec + 2;
}
}
}
osip_message_free(msg);
return sip;
}
